Call drop after 30+sec happens if RTP is not received by asterisk for 30
seconds (RTP Timeout).
You should look for media IP address in SDP. If there is firewall, apart
from port UDP/5060, you also need to open port UDP/1-UDP/2
(standard RTP ports)
You should try with RTP debug. It shoul
On 22.11.2014, at 13:40, Yves A. wrote:
> I have a really strange problem which is driving me crazy for days now.
>
> If I register my asterisk (tried all versions from 1.6 up to 13.x) with one
> sip registrar,
> everything works... calls go out and call come in... no 32 seconds limit.
>
> but
t;mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Yves A.
Sent: Saturday, November 22, 2014 8:06 AM
To: asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but
only when
A
M
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but
only when
Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
but as soon as I configure another sip registration on another server,
outgoing
calls drop after 32 seconds.
Are both your servers
s.digium.com
Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when
Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
but as soon as I configure another sip registration on another server,
outgoing
calls drop after 32 seconds.
Are both your servers behind the same NAT
ctmedia=no in sip.conf.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
> Sent: Saturday, November 22, 2014 8:06 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [aster
call drops after 32 seconds, but only when
Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
>> but as soon as I configure another sip registration on another server,
>> outgoing
>> calls drop after 32 seconds.
> Are both your servers behind the same NAT router?
>
thanks f
Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
but as soon as I configure another sip registration on another server,
outgoing
calls drop after 32 seconds.
Are both your servers behind the same NAT router?
thanks for taking part...
I don´t know...
one is
siptrunk.ovh.net
and the other one
> but as soon as I configure another sip registration on another server,
> outgoing
> calls drop after 32 seconds.
Are both your servers behind the same NAT router?
--
Andreas Sikkema
--
_
-- Bandwidth and Colocation Provide
hi,
I have a really strange problem which is driving me crazy for days now.
If I register my asterisk (tried all versions from 1.6 up to 13.x) with
one sip registrar,
everything works... calls go out and call come in... no 32 seconds limit.
but as soon as I configure another sip registration
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